search for: deltathree

...n message ( with Autherization section, and that get accepted by the destination host). Why is this ? Isnt there a way to tell * to send with Autothorization message the first attempt ? Asterisk sends this first 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipauth.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a From: <sip:1510xxxxxxx@sipauth.deltathree.com>;tag=as3e96887d To: <sip:1510xxxxxxx@sipauth.deltathree.com> Call-ID: 1dfb33d406292f5327646eae0683e11b@66.121.81.25 CSeq: 106 REGISTER User-Agent: Asterisk PBX Expires:...

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( T...

...blem. Any leads will be much appreciated. Attached are the conf files and logs #SIP.CONF ; SIP Configuration for Asterisk [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = from-sip register => mylogin:mypassword@sipauth.deltathree.com/2905 [iconnect] type=friend username=mylogin password=mypassword host=natrelay.deltathree.com #EXTENSIONS.conf [general] static=yes writeprotect=no [globals] [intern] include => iconnecthere [iconnecthere] ;Outbound calls to PSTN numbers via iconnecthere.com's SIP service exten =&gt...

Hi, i have diffirent provider example(3 single account in deltathree, 4 account in packet8 and so on) . How this possible to make the three individual sip account in deltathree act as trunk so that i cannot get a busy call. If line one fail goto line 2 then line 3 or another trunk line 1 then line 2 then line3....I read it in asterisk at home but the script...

...f file recently. Here is what I have: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw tos=lowdelay tos=185 ;register= xxxxxxxx:xxxx@sipauth.deltathree.com ;register= xxxxxxxx:xxxx@sipauth.deltathree.com [iconnect] type=friend username=xxxxxxxx password=xxxx host=sipauth.deltathree.com ;host=213.137.73.178 canreinvite=no Has anybody experienced this before?

...debug on, I can see that something is happening everytime a call is received from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would appreciate any help on this. Thanks Below is my sip file register=442087926805:somepassword@sipauth.deltathree.com:5060 [iconnecthere] type=friend secret=somepassword username=11232634 host=sipauth.deltathree.com canreinvite=no ;nat=yes context=default ;dtmfmode=inband disallow=all ;allow=all allow=gsm allow=ulaw allow=alaw allow=g726 allow=g723 This is the sip debug info when a call comes in from iconnec...

...iguration files: (commented lines represent options I tried). n= my Iconnect phone number p=Iconnect password u=Iconnect User id sip.conf [general] port = 5060 ;bindaddr = 0.0.0.0 ;bindaddr = 212.199.7.106 context = from-sip ;callerid=No CallID ;register=nnnnnnnnnnn:pppp@sipauth.deltathree.com/ nnnnnnnnnnn ;register= uuuuuuuu:pppp @natrelay.deltathree.com/phone ;register=uuuuuuuu:pppp@sipauth.deltathree.com/phone [iconnect] type=friend secret=pppp username=uuuuuuuu host=sipauth.deltathree.com ;host=natrelay.deltathree.com ;fromdomain=xxx.xxx.xxx.xxx ;dtmfmode=inband nat...

...e 480 "Temporarily not available" back from 213.137.73.176.. i had a look on google and aparantly if you change the codec it fixed it or something.. i tried all of them nothing seemed to work. heres a copy of my sip.conf [iconnect] type=friend username=xxxxxxxx secret=xxxx host=sipauth.deltathree.com qualify=1000 callerid="1111178197026" allow=g721.1 and for the xlite [2001] type=friend callerid="JUSTIN XLITE" <2001> username=2001 secret=2001 context=-phones mailbox=2001 host=dynamic nat=no qualify=yes allow=9721.1 ive been playing around with the nat settings...

Hi, I was able to connect asterisk to iconnect's service. It took me almost two hours, but it's because I was having NAT trouble. I finally discovered that you can set the iconnect host to natrealy.deltathree.com to make it work. (for those of you who, like me, don't have the time to search the archive I'll provide a working sample in a minute) My problem was sound quality. It was horrible. It took as much as five seconds to bridge iconnect to my sip soft phone, and then the voice record...

...ip.conf [general] ;calls arrive from sip lands here port=5060 context=default-in disallow=all allow=ulaw allow=alaw allow=g729 maxexpirey=180 defaultexpirey=160 ;Connect to Free World Dialup (no NAT) register=61358:xxxxxxx@fwd.pulver.com/61358 ;Connect to iconnect register=15108688610:xxxx@sipauth.deltathree.com/15108688610 canreinvite=no [iconnect] ;incoming does not land here, why ? outgoing is fine type=friend secret=xxxx username=yyyyyyyy host=sipauth.deltathree.com dtmfmode=inband ; required by iconnect context=iconnect-in canreinvite=no allow=alaw allow=ulaw allow=g729 [fwd] ;incoming does n...

...ocess): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames Here is my sip.conf [general] port=5060 context=default allow=g723.1 maxexpirey=180 defaultexpirey=160 ;Connect to iconnect register=1510xxxxxx:xxxx@natrelay.deltathree.com/1510xxxxxx [iconnect] type=friend secret=xxxx username=xxxxxxx host=natrelay.deltathree.com dtmfmode=inband canreinvite=no context=vobb-in allow=g723.1 Can someone be able to debug this ? If I make the codec to g729, call not even get through. * complains that it can't bridge the codec....

...User/ANR Call ID Seq (Tx/Rx) Format 213.137.73.140 133357xx 1e7ff5217c3 00104/00000 UNKN (d) 1 active SIP channel(s) ast1000*CLI> sip show history 1e7ff5217c3dbd3d737b8ddc6ceaf087@192.168.254.250 ast1000*CLI> * SIP Call 1. TxReqRel REGISTER sip:sipauth.deltathree.com SIP/2.0 / 104 REGISTER 2. Rx SIP/2.0 100 Trying / 104 REGISTER However, I seem to be successfully registered: ast1000*CLI> sip show registry Host Username Refresh State sipauth.de...

...ing to register a SIP service(s) with FWD and ICONNECTHERE. Here is my sip.conf: ------------------------------------------------ [general] port =5060 bindaddr =0.0.0.0 context =default calledid=No CallID register => 400703:PASSWORD@fwd.pulver.com/400703 register => USERNAME:PASSWORD@sipauth.deltathree.com/400703 [fwd] type=friend username=USERNAME secret=PASSWORD host=fwd.pulver.com dtmfmode=inband [iconnect] type=friend username=USERNAME secret=PASSWORD host=sipauth.deltathree.com dtmfmode=inband ---------------------------------------------- Here is error output i get: File chan_sip.c, Line...

...guration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw [iconnect] type=friend username=******** password=**** host=sipauth.deltathree.com ;host=213.137.73.178 All I have been able to find about this topic is that 19 is supposed to be comfort noise (whatever that is) Any help is appreciated

...isk setup is talking with iconnecthere. My sip.conf is setup to disallow=all, allow=gsm, allow=ulaw, allow=alaw, however it doesn't make a difference if I connect with either codec. Here is my iconnecthere section with user/pass removed: [iconnect] type=friend secret= username= host=natrelay.deltathree.com dtmfmode=inband canreinvite=no All this seems pretty standard. What settings or debug information should I look at to fix this issue. Has anyone else encountered this problem before. I'd be happy to supply more information about my setup but since I'm not sure where the problem woul...

...different symptoms for inbound calls from iconnect. See my previous email. In short, they INVITE me to the call, but don't seem to hear my replies and then dump the call into voicemail after timing out. This is not on NAT; this is all clear-IP transactions. If anyone from iconnecthere or deltathree is lurking on the list, please see if you can figure out what is fubar'ed there, if anything. JT

...ly different question. At the current CVS can the * SIP client register with a SIP provider (such as ICH) when the * box itself is behind a NAT. I have an ICH account which I can configure fine on an ATA 186 using their "NAT instructions" . Can the SIP client on * be set to use natrelay.deltathree.com as the gateway and/or outbound proxy ? Any info appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030506/faf17adf/attachment.htm

...I believe that digest authentication can be end-to-end or proxy-to-proxy and this is my problem??( maybe not) I have tried to use the type=friend / peer , and other combinations of directives, in sip.conf to no avail. Looking at this list, it seems people are able to authenticate and use the deltathree with asterisk. Can someone please shed some light to what might be going on, or how to specifiy the digest behavior to be proxy-to-proxy? Thanks in advance, -Eric ------------------------------------------------------------------------ ---------------- This email is composed of >80%...

Hi All, I can only seem to get iconnecthere working with incoming calls intermittently. One minute it seems to work, and the next it doesn't. I am not aware of anything being changed in the config files. Outgoing calls work ok all the time. The Asterisk box is behind NAT so that does complicate things slightly. However, the Iconnecthere PCPhone client software works perfectly for

hi i got iconnect to work, works pretty well now except calls sometimes (more often than not) hang up after a couple of minutes.. heres a bit of the debuging Record-Route: <sip:61892142222@213.137.73.178:5060;maddr=213.137.73.176> From: sip:61892142222@natrelay.deltathree.com;tag=3281050172-73809 To: "JUSTIN XLITE" <sip:2001@61.95.68.84>;tag=as09766a78 Call-ID: 5df848b10eea268512db62e41181dc3d@61.95.68.84 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2001@61.95.68.84> Content-Length: 0 to...